Don't worry about it Moises. I'll tell you all issues I could find... 2010/12/5 Moises Silva <moises.silva@gmail.com>
2010/12/4 Juan Antonio Ibañez Santorum <juanito1982@gmail.com>
You are true. Last git version works as expected.
Thank you very much
We're very interested in hearing about this kind of inter-networking issues between SIP and the PSTN, if you find any other glitch, let us know.
Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy@sangoma.com
2010/12/2 David Yat Sin <dyatsin@sangoma.com>
Hi Juan Antonio,
Can you pull the latest Freeswitch git. This should be fixed now.
David
*David Yat Sin, BEng* Software Engineer Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
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On 11/24/2010 1:38 PM, Juan Antonio Ibañez Santorum wrote:
Any news?
2010/11/18 David Yat Sin <dyatsin@sangoma.com>
Hi Juan Antonio, We are looking into this and will get back to you in a day or two.
David
*David Yat Sin, BEng* Software Engineer Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
t. +1 800 388 2475 x119 t. +1 905 474 1990 x119 f. +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>| Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>| Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>| Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>| Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>| Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>| Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
On 11/18/2010 3:50 PM, Juan Antonio Ibañez Santorum wrote:
Hello boys!
I have one problem after updating from openzap to freetdm. Now, whe FS receives 'Proceeding' from PRI sends a SIP 180 which makes the phone to play a fake ring tone. After some seconds, when FS receives 'Alerting' from PRI, FS sends a SIP 183 with the real ring tone.
Is there any way to avoid first SIP 180? If the calle is busy, I can hear a ringing tone between SIP 180 and SIP 183 and a busy tone after SIP 183. Same if calle is a mobile phone and it is out of signal.
Regards
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